[..] |
ASTERISK-23004: Documentation: improve documentation of pjsip endpoints behind NAT |
ASTERISK-23006: Fake NOTIFY in blind call transfer |
ASTERISK-23007: Asterisk Swagger: bump the version number |
ASTERISK-23008: Local channels loose CALLERID name when DAHDI channel connects |
ASTERISK-23010: No BYE message sent when sip INVITE is received |
ASTERISK-23011: [patch]configure.ac and pbx_lua don't support lua 5.2 |
ASTERISK-23012: crash in pjsip_transport_dec_ref when called from rx_task_data_destroy in res_pjsip_registrar |
ASTERISK-23013: [patch] Deadlock between 'sip show channels' command and attended transfer handling |
ASTERISK-23017: Crash on inbound calls using WebRTC config with ICE Servers -signal 6 abort, while in ice_worker_thread |
ASTERISK-23018: PJSip 'allow=all' results in failed SDP negotiation |
ASTERISK-23020: PJSip - Multihomed machine returning wrong IP address |
ASTERISK-23021: Typos in code : "avaliable" instead of "available" |
ASTERISK-23025: Asterisk Crashing |
ASTERISK-23026: [patch]Asterisk should send STUN messages using role 'ICE-CONTROLLING' when being offerer of SDP |
ASTERISK-23027: [patch] Spelling typo "transfered" instead of "transferred" |
ASTERISK-23028: [patch] Asterisk man pages contains unquoted minus signs |
ASTERISK-23029: Webrtc + realtime. Dial to webrtc clients only with rtcachefriends=yes |
ASTERISK-23030: MixMonitor/StopMixMonitor AMI commands not present in Asterisk 1.8.24 |
ASTERISK-23031: CLONE - Asterisk Crashing |
ASTERISK-23032: Example Configurations use old '=>' syntax |
ASTERISK-23033: direct call can't be placed over h323 if peer is not registered in gatekeeper |
ASTERISK-23034: [patch] manager Originate doesn't abort on failed format_cap allocation |
ASTERISK-23035: ConfBridge with name longer than max (32 chars) results in several bridges with same conf_name |
ASTERISK-23037: res_pjsip/distribute: crash when distributing transaction during a mutex unlock in pjsip |
ASTERISK-23038: Need config option to enable PJSIP logger at load time |
ASTERISK-23046: Custom CDR fields set during a GoSUB called from app_queue are not inserted |
ASTERISK-23047: Orphaned (stuck) channel occurs during a failed SIP transfer to parking space |
ASTERISK-23048: Simple chan_pjsip sip to sip call, call dies about 30 seconds in |
ASTERISK-23049: BridgeWait: Attended transfer to BridgeWait bridge can fail if the transferer is the only channel in the bridge |
ASTERISK-23050: 'pjsip list contacts' results in crash |
ASTERISK-23051: ARI: channel variables in JSON breaks passing parameters in JSON |
ASTERISK-23053: The users of ao2_iterator_cleanup() are violating the ao2_iterator opacity. |
ASTERISK-23054: [patch] ast_custom_escalating_function allocation's leak |
ASTERISK-23056: [patch]INFINITY and NAN undefined |
ASTERISK-23061: [Patch] 'textsupport' setting not mentioned in sip.conf.sample |
ASTERISK-23062: res_pjsip AOR config option qualify_frequency is inconsistently respected |
ASTERISK-23064: res_pjsip_pubsub: crash - pjlib function called from external thread |
ASTERISK-23065: On Asterisk start, device state is INVALID for previously registered PJSIP endpoints, despite re-registrations |
ASTERISK-23068: http: Implement support for chunked Transfer-Encoding |
ASTERISK-23069: Custom CDR variable not recorded when set in macro called from app_queue |
ASTERISK-23070: Memory leak when using CLI through AMI |
ASTERISK-23071: pjsip: mailboxes documentation is lacking |
ASTERISK-23072: MWI subscription from Cisco SPA fails with PJSIP |
ASTERISK-23073: Asterisk crashes randomly when using chan_unistim |
ASTERISK-23074: Crash in ChanIsAvail app |
ASTERISK-23081: PJSip Tab Expansion erroring |
ASTERISK-23082: Including g722 in pjsip codec configuration results in unexpected SDP offers |
ASTERISK-23083: PJSip not honouring rtp port definitions |
ASTERISK-23084: [patch]rasterisk needlessly prints the AST-2013-007 warning |
ASTERISK-23086: pjsip aor record name must match username for successful registration |
ASTERISK-23087: PJSIP cant handle SUBSCRIBE with Event header dialog instead of presence |
ASTERISK-23090: chan_sip fails to transmit BYE request to WebSocket connected peer after a failed attended transfer |
ASTERISK-23091: possible memory leak in abstract jitterbuffer |
ASTERISK-23092: cli: pjsip show endpoint <endpoint> shows allow/disallow codecs the same |
ASTERISK-23097: Module res_ldap uses cn for peer username regardless of attribute |
ASTERISK-23098: [patch]possible null pointer dereference in format.c |
ASTERISK-23099: [patch] WSS: enable ast_websocket_read() function to read the whole available data at first and then wait for any fragmented packets |
ASTERISK-23100: [patch] In chan_mgcp the ident in transmitted request and request queue may differ - fix for locking |
ASTERISK-23101: pjsip: crash when parsing scheme from SIP URI |
ASTERISK-23102: sorcery: crash when reloading res_pjsip outbound registration |
ASTERISK-23103: [patch]Crash in ast_format_cmp, in ao2_find |
ASTERISK-23104: Specifying the SetVar AMI without a Channel cause Asterisk to crash |
ASTERISK-23105: Asterisk crashed in 'inchar' at app_voicemail.c when performing forward |
ASTERISK-23106: pjsip: ACK to 200 OK sent to private IP address on outbound channel's INVITE request |
ASTERISK-23108: “rtcachefriends”, WebRTC, and Realtime |
ASTERISK-23111: astdb entries added just before shutdown are not saved |
ASTERISK-23114: Formats: Improve performance of Asterisk by handling formats in a more performant manner |
ASTERISK-23119: [patch] main/bridging.c sometimes fails to pthread_join the bridge thread |
ASTERISK-23120: ARI/AMI: allow objects created via interfaces to have their unique ID specified by the external application |
ASTERISK-23125: ARI: URI is case sensitive |
ASTERISK-23126: res_pjsip_messaging: Asterisk should make use of the P-Asserted-Identity headers with regard to in/outbound MESSAGE requests |
ASTERISK-23127: feature code transfer - transferred channel's CDR(uniqueid) is updated with uniqueid from the transferor's channel |
ASTERISK-23128: res_ari: Memory leak on response headers and some JSON response messages |
ASTERISK-23129: segfault in res_pjsip_pubsub.so |
ASTERISK-23130: Transfer of parked call causes 100% CPU and orphaned channels |
ASTERISK-23133: Documentation fix - MASTER_CHANNEL Unexpected Behaviour |
ASTERISK-23134: [patch] res_rtp_asterisk port selection cannot handle selinux port restrictions |
ASTERISK-23135: Crash - segfault in ast_channel_hangupcause_set - probably introduced in 11.7.0 |
ASTERISK-23136: cel.conf is read by more than one core module. |
ASTERISK-23137: Queue is not playing MOH |
ASTERISK-23138: queue_log agent is not consistent |
ASTERISK-23139: Security: Remote crash in res_pjsip_exten_state |
ASTERISK-23141: Asterisk crashes on Dial(), in pbx_find_extension at pbx.c |
ASTERISK-23142: Large timestamp skew in RTP stream during blind transfer |
ASTERISK-23143: ARI: subscribing to an already subscribed resource returns a 500 error |
ASTERISK-23145: Sporadic one way audio between a SIP hard phone and a SIPML5 browser client on LAN |
ASTERISK-23147: [patch] main/format.c: find_interface segfault during shutdown |
ASTERISK-23154: Manager: ExtensionStatus event does not present information in a human readable way |
ASTERISK-23160: Fax CNG detection when no apparent CNG present in inbound audio stream |
ASTERISK-23161: how to connect voip phones using static IP address with asterisk server? |
ASTERISK-23162: SQLite3 CDRs are not being populated with CDR variables |
ASTERISK-23164: CDRs: mid-call/pre-dial handlers perturb context/exten/app/data fields during Dial |
ASTERISK-23166: T.38 Fax outgoing call via chan_ooh323 breaks after rtptimeout timer |
ASTERISK-23168: Overriding outbound_auth in a pjsip registration causes ERROR, assert failure. |
ASTERISK-23169: [patch] Support getting local SIP IP using CHANNEL() dial plan function |
ASTERISK-23171: Crash in res_rtp_asterisk on WebRTC incoming call |
ASTERISK-23172: PJSip missing functionality provided by channel variable: SIP_URI_OPTIONS |
ASTERISK-23173: PJSip missing feature: SIPPEER |
ASTERISK-23174: CDR documentation issues |
ASTERISK-23176: WebRTC missing ice-ufrag and ice-pwd compiling with BETTER_BACKTRACES and DONT_OPTIMIZE |
ASTERISK-23177: [patch] RealTime cant update sipbuddies table when registering or updating friend |
ASTERISK-23178: devicestate.h: device state setting functions are documented with the wrong return values |
ASTERISK-23179: [patch] Formats: allow improved performance in Asterisk 11 |
ASTERISK-23181: Segmentation fault core dumped EVERYTIME hold button pressed on Bria Android when using MulticastRTP |
ASTERISK-23182: SIP/OK response to a SIP/BYE request is sent to the wrong port |
ASTERISK-23184: [patch] Cosmetic error when cross-compiling as ./configure calls ./menuselect/configure |
ASTERISK-23185: qualify=yes does not respect outboundproxy setting |
ASTERISK-23186: [patch] Add usegmtime option to cel_pgsql |
ASTERISK-23187: testsuite: Write a test for the AstDB |
ASTERISK-23190: WebRTC (WSS + TLS) No Audio |
ASTERISK-23191: Cannot get local SDP on incoming calls (WebRTC) |
ASTERISK-23193: res_rtp_asterisk: Crash in pj_ice_sess_on_rx_pkt when handling RTCP packet |
ASTERISK-23194: Segfault when loading module func_enum |
ASTERISK-23195: Music on hold does not works at all, when using Queue application |
ASTERISK-23196: loss of voice during a call |
ASTERISK-23197: Problem with quick dial of dtmf signals |
ASTERISK-23198: testsuite: Write a test for marked user/unmarked user interaction |
ASTERISK-23204: Device state cache requires improvement |
ASTERISK-23207: Crash on shutdown in ccss due to scheduler context already being destroyed in cc_generic_monitor_cancel_available_timer |
ASTERISK-23210: Security: Remote crash in res_pjsip. |
ASTERISK-23211: auto mute in confbridge |
ASTERISK-23213: SIP over WS: Audio problems when upgrading to 11.8 from 11.7 with endpoints behind NAT |
ASTERISK-23214: chan_sip WARNING message 'We are requesting SRTP for audio, but they responded without it' is ambiguous and wrong in some cases |
ASTERISK-23217: testsuite: Write a test for call files |
ASTERISK-23218: testsuite: Write a test for call file retries |
ASTERISK-23219: Asterisk restarts with increasing LA |
ASTERISK-23220: STACK_PEEK function with no arguments causes crash/core dump |
ASTERISK-23221: Asterisk Crash on 32bit Linux with MALLOC_DEBUG enabled |
ASTERISK-23225: Asterisk crash while placing a call (signal 6 - Aborted) |
ASTERISK-23227: Both Digium Phones and Softphones no longer registered with Asterisk PJSIP |
ASTERISK-23229: testsuite: Write basic tests for AgentLogin |
ASTERISK-23230: testsuite: Write a nominal AgentLogin/AgentRequest test |
ASTERISK-23231: Since 405693 If we have res_fax.conf file set to minrate=2400, then res_fax refuse to load |
ASTERISK-23232: LocalBridge AMI Event LocalOptimization value is opposite to what's expected |
ASTERISK-23233: alembic missing scripts for certain realtime tables |
ASTERISK-23235: pjsip transport/tos interpreted differently than endpoint/tos_audio |
ASTERISK-23242: AGISTATUS returns false statuses for some scripts when using AGI(script.agi) |
ASTERISK-23243: Messag(body) starting with one space not displayed in clients. |
ASTERISK-23244: [patch] testsuite: run-local fails to recognize modules |
ASTERISK-23245: tls not working |
ASTERISK-23246: DEBUG messages in sdp_crypto.c display despite a DEBUG level of zero |
ASTERISK-23248: Asterisk 1.8.23.0 Crashes Signal 11/Segfault |
ASTERISK-23249: Skinny subchannel locking issues |
ASTERISK-23250: CDR(start) function is broken due to sizeof dereference |
ASTERISK-23251: chan_sip - RTP Packetization set in general section not applied when Dialing direct to a SIP URI |
ASTERISK-23254: Bad ao2_find() usage in pjsip_options.c |
ASTERISK-23255: UUID included for Redhat, but missing for Debian distros in install_prereq script |
ASTERISK-23257: sip.conf option pendantic=yes not working for dialed numbers with sign # |
ASTERISK-23258: Target_uri for LiveRecording model in ARI |
ASTERISK-23259: Local channel agent is unavailable for 5 seconds after login |
ASTERISK-23260: [patch]ForkCDR v option does not keep CDR variables for subsequent records |
ASTERISK-23261: [patch]Output mixup in ${CHANNEL(rtpqos,audio,all)} |
ASTERISK-23262: Audio degredation with codec_dahdi and ChanSpy'ing |
ASTERISK-23263: testsuite: Write tests for AgentLogin/AgentRequest that covers acknowledgement |
ASTERISK-23265: Preloading Certain Modules in Asterisk 12 causes a core dump |
ASTERISK-23266: [patch]pjsip_cli: Memory leak in ast_sip_cli_print_sorcery_objectset |
ASTERISK-23267: Expires In Contact Header Of 200 OK From Outbound Registration Response Is Ignored Due To Contact URI Parameters Being Added |
ASTERISK-23268: AgentRequest waits 2 minutes for agent to acknowledge call |
ASTERISK-23274: [patch] Manager action QueueStatus doesn't show recent realtime changes |
ASTERISK-23275: CLI command 'pjsip show registrations' missing |
ASTERISK-23276: Look at adding the 'pjsip show channel' command |
ASTERISK-23279: [patch]Asterisk doesn't support the dynamic payload change in rtp mapping in the 200 OK response |
ASTERISK-23280: Failed bridge when running ParkAndAnnounce from macro initiated from applicationmap |
ASTERISK-23282: Documentation - Tab completion and CLI usage documentation do not indicate that 'all' is accepted for 'confbridge kick all' |
ASTERISK-23283: SipNotify Action over AMI do not allow multiline content with \n |
ASTERISK-23285: PJSIP: Correct MWI behavior when mailboxes are specified for an endpoint and AOR |
ASTERISK-23287: res_pjsip_refer: Crash during attended transfer when attended->transferer_second channel is NULL |
ASTERISK-23289: Application Record turning off after 5 minutes |
ASTERISK-23290: chan_sip: ast_bridge_transfer_blind causes channel to be hung up immediately, leading to BYE request being sent before NOTIFY |
ASTERISK-23293: Testsuite asttest doesn't compile with Lua 5.2 |
ASTERISK-23294: SIPqualifypeer (over AMI) does not update status when peer is UNREACHABLE |
ASTERISK-23295: ARI: ChannelEnteredBridge event not delivered to client during bridge move operation |
ASTERISK-23297: Asterisk 12, pbx_config.so segfaults if res_parking.so is not loaded, or if res_parking.conf has no configuration |
ASTERISK-23298: testsuite: Write a test for waitmarked users only |
ASTERISK-23300: testsuite: Write a test for waitmarked/normal user interaction |
ASTERISK-23301: testsuite: Write a test for waitmarked/marked/normal user interaction |
ASTERISK-23302: testsuite: Write a test for end_marked/wait_marked/marked users |
ASTERISK-23306: chan_sip: Asterisk creates ACK with empty Route: headers |
ASTERISK-23308: [patch] Unable to specify non-standard destination port if "callbackextension" enabled |
ASTERISK-23309: [patch] Failed to pause/unpause already paused/unpaused realtime queue member |
ASTERISK-23310: bridged channel crashes in bridge_p2p_rtp_write |
ASTERISK-23311: Manager - MoH Stop Event fails to show up when leaving Conference |
ASTERISK-23313: ACK sent to wrong destination in CANCEL dialog |
ASTERISK-23314: Documentation - Various Caller ID and COLP related function arguments need some clarification |
ASTERISK-23315: Asterisk 12 rc1 doesn't compile on my raspberry |
ASTERISK-23317: CLONE - ACK sent to wrong destination in CANCEL dialog |
ASTERISK-23318: CLONE - ACK sent to wrong destination in CANCEL dialog |
ASTERISK-23319: Segmentation fault in queue_exec at app_queue.c |
ASTERISK-23320: Preloading pbx_config.so with a CustomPresence hint defined results in crash |
ASTERISK-23322: Unable to use SIP INVITE authentication with type=peer and device name mismatch with username |
ASTERISK-23324: [patch] - QLOOG commiting Japanese translated prompts |
ASTERISK-23325: Asterisk is Running but not processing anything |
ASTERISK-23328: Asterisk crash in ast_cdr_setapp() at cdr.c |
ASTERISK-23329: Realtime Peers behind NAT becomes unreachable after peer reload |
ASTERISK-23336: Asterisk warning "Don't know how to indicate condition 33 on ooh323c" on outgoing calls from H323 to SIP peer |
ASTERISK-23337: One way audio issues with WebRTC - PJNATH limits number of ICE candidates |
ASTERISK-23338: how to solve this error during call parking:bad magic number |
ASTERISK-23339: Segfault in __ao2_find at astobj2.c, in find_interface at format.c |
ASTERISK-23340: Security Vulnerability: stack allocation of cookie headers in loop allows for unauthenticated remote denial of service attack |
ASTERISK-23342: PJSIP Testing: Subscription tests |
ASTERISK-23343: PJSIP Testing: moar MWI tests |
ASTERISK-23344: PJSIP Testing - moar extension state tests |
ASTERISK-23348: Allocation failure in ast_unreal_new_channels causes core dump |
ASTERISK-23349: Testsuite: manager live_dangerously tests are missing a parameter for on_failure |
ASTERISK-23350: Testsuite: manager/acl-login fails on some platforms |
ASTERISK-23351: [patch]Updating realtime sippeers using res_config_pgsql backend fails when 'port' column is null |
ASTERISK-23352: Testsuite: tests/cdr/originate-cdr-disposition fails, reports success |
ASTERISK-23353: testsuite: Write a test for ChannelRedirect |
ASTERISK-23354: testsuite: write a test for the BridgeWait application's S option |
ASTERISK-23355: testsuite: write a test for the BridgeWait application's e options |
ASTERISK-23356: testsuite: write a test that tests for an announcer channel/participant channel roles |
ASTERISK-23365: [patch]Testsuite: chan_sip rfc2833_dtmf_detect fails for non-root (rawsocket access denied) |
ASTERISK-23369: [patch] Testsuite: Timeout shutting down asterisk should result in failed test |
ASTERISK-23370: Wiki Documentation - Getting Started/Beginning Asterisk |
ASTERISK-23371: Wiki Documentation - Getting Started/Installing Asterisk/Installing Asterisk from Source |
ASTERISK-23372: Wiki Documentation - Getting Started/Hello World |
ASTERISK-23373: [patch]Security: Open FD exhaustion with chan_sip Session-Timers |
ASTERISK-23374: Wiki Documentation - Getting Started/Basic Configuration |
ASTERISK-23375: Wiki Documentation - Operation/System Requirements |
ASTERISK-23377: MixMonitor Append Flag Ignored |
ASTERISK-23378: [patch]Queue with 'ringinuse=no' and members in realtime can get several calls at the same time (with patch) |
ASTERISK-23381: [patch]ChanSpy- Barge only works on the initial 'spy', if the spied-on channel makes a new call, unable to barge. |
ASTERISK-23382: [patch]Build System: make -qp can corrupt menuselect-tree and related files |
ASTERISK-23383: Wrong sense test on stat return code causes unchanged config check to break with include files. |
ASTERISK-23389: Asterisk 1.8.23.0 Crashes Signal 11/Segfault (app_queue QueueRemove Address Out of Bounds) |
ASTERISK-23390: NewExten Event with application AGI shows up before and after AGI runs |
ASTERISK-23391: Audit dialplan function usage of channel variable |
ASTERISK-23395: Wiki Documentation - Operation/Running Asterisk |
ASTERISK-23396: Wiki Documentation - Operation/Maintenance and Upgrades |
ASTERISK-23397: AMI Park when not specifying TimeoutChannel causes announcement and call back to channel specified in Channel argument |
ASTERISK-23398: Wiki Documentation - Operation/Logging |
ASTERISK-23399: Wiki Documentation - Operation/Asterisk Command Line Interface |
ASTERISK-23404: Patch to pjproject to remove (most of) third_party directory |
ASTERISK-23406: [patch]Fix typo in "sip show peer" |
ASTERISK-23407: Fix the FSF address in the headers of lots of pjproject files |
ASTERISK-23410: Asterisk not executing properly on CentOS 6.5 |
ASTERISK-23411: Asterisk segfault sip_get_codec |
ASTERISK-23412: Theoretical ref-leak in chan_sip autokill |
ASTERISK-23414: chan_sip stops responding after a about 5-20 minutes |
ASTERISK-23415: Ignore me |
ASTERISK-23416: stasis app: ChannelCreated event not being received |
ASTERISK-23418: ast_read() on chan '...' called with no recorded file descriptor. |
ASTERISK-23419: [patch]Crash, segfault due to attempt to read an invalid rtp instance |
ASTERISK-23420: [patch]Memory leak in manager_add_filter function in manager.c |
ASTERISK-23421: AMI: more metadata on voicemail |
ASTERISK-23422: Wiki Documentation - Fundamentals/Asterisk Architecture |
ASTERISK-23424: Dependency in libsrtp.so included with pjproject breaks build system |
ASTERISK-23425: No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd. |
ASTERISK-23426: sip show peers became empty in realtime after reload or sip reload |
ASTERISK-23427: tests/callparking: Crash on assert when obtaining channel information from Stasis cache during bridge leave event |
ASTERISK-23433: ARI: Add 'tones' as a URI scheme for /play operations on resources that support media (bridges, channels) |
ASTERISK-23435: PJSIP: Fix the DNS resolution (whoops) |
ASTERISK-23437: ARI: Add the ability to add properties to a bridge on creation |
ASTERISK-23438: Wiki Documentation - Fundamentals/Asterisk Configuration/Asterisk Configuration Files |
ASTERISK-23439: Wiki Documentation - Fundamentals/Asterisk Configuration/Database Support Configuration |
ASTERISK-23442: Wiki Documentation - Fundamentals/Asterisk Configuration/Sorcery |
ASTERISK-23444: Playback and Record events not subscribed to when calling Play/Record on bridge |
ASTERISK-23446: PJSIP Testing: Nominal blind transfers (Caller initiated) |
ASTERISK-23447: PJSIP Testing: Nominal blind transfers (Callee initiated) |
ASTERISK-23451: res_pjsip_dtmf_info: Create DTMF INFO request tests |
ASTERISK-23452: Notification when no message left |
ASTERISK-23453: res_pjsip_header_funcs: Add tests for PJSIP_HEADER |
ASTERISK-23456: RES XMPP |
ASTERISK-23457: SQlite3: Realtime queue loading fails after PRAGMA query result |
ASTERISK-23459: [patch]Incorrect check for key field and NULL column values in update_odbc |
ASTERISK-23460: ooh323 channel stuck if call is placed directly and gatekeeper is not available |
ASTERISK-23461: Only first user is muted when joining confbridge with 'startmuted=yes' |
ASTERISK-23462: Cannot disable SIP debugging via CLI after enabling with conf file option - also 'sip set debug off' reports debugging disabled, when it really isn't |
ASTERISK-23468: Wiki Documentation - Fundamentals/Key Concepts/Channels |
ASTERISK-23469: Wiki Documentation - Fundamentals/Key Concepts/Bridges |
ASTERISK-23470: Wiki Documentation - Fundamentals/Key Concepts/Frames |
ASTERISK-23471: Wiki Documentation - Configuration/Core Configuration |
ASTERISK-23472: Wiki Documentation - Configuration/Channel Drivers/SIP |
ASTERISK-23474: Wiki Documentation - Configuration/Channel Drivers/DAHDI |
ASTERISK-23475: res_odbc crash |
ASTERISK-23484: HANGUPCAUSE can't get any information on a hungup inbound channel unless called with no arguments, plus unaffected by use_q850_reason |
ASTERISK-23485: Asterisk suddenly stops writing data to astdb |
ASTERISK-23486: Wiki Documentation - Configuration/Channel Drivers/Local Channel |
ASTERISK-23487: features.conf cant load from realtime because features_config.c starts before loader.c |
ASTERISK-23488: Logic error in callerid checksum processing |
ASTERISK-23489: Vulnerability in res_pjsip_pubsub: unauthenticated remote crash in during MWI unsubscribe without being subscribed |
ASTERISK-23491: Asterisk-ARI-not updated after reload command |
ASTERISK-23492: Add option to safe_asterisk to disable backgrounding |
ASTERISK-23493: SIP Attended Transfer CDR record has differing linkedid than associated CDRs from the entire call - conflicts with spec |
ASTERISK-23496: SIP Channel Stops Accepting Packets |
ASTERISK-23497: chan_sip SIP protocol attended transfer, with directmedia=yes results in a simple bridge, typically with no audio |
ASTERISK-23498: Asterisk PJSIP transport configuration fails on parsing of 'cipher' option, any valid option is reported as unsupported |
ASTERISK-23499: app_agent_pool: Interval hook prevents channel from being hung up |
ASTERISK-23501: Copy 'Referred-By' header to outgoing INVITE |
ASTERISK-23502: Channel variable SIPREFERTOHDR not being set during blind transfer |
ASTERISK-23503: Using AMI via HTTP adds extra characters (: and space) to some request headers. |
ASTERISK-23504: [patch] MESSAGE string is not present in allowed methods in SIP header |
ASTERISK-23508: Memory Corruption in __ast_string_field_ptr_build_va |
ASTERISK-23509: [patch]SayNumber for Polish language tries to play empty files for numbers divisible by 100 |
ASTERISK-23510: JABBER_STATUS fails with improper code 7 for unavailable clients |
ASTERISK-23511: Asterisk-12.1.1 segmentation fault |
ASTERISK-23512: Inaccurate comment in manager.conf.sample |
ASTERISK-23514: The pjsip.conf aor qualify contact parameters are not updated on reload. |
ASTERISK-23515: ICE info missing ice-ufrag and ice-pwd in INVITE received by caller |
ASTERISK-23520: testsuite: Write a basic channel Pickup application test |
ASTERISK-23521: testsuite: nominal DISA tests: authentication |
ASTERISK-23522: testsuite: nominal DISA tests: no authentication |
ASTERISK-23523: testsuite: nominal DISA tests: no context |
ASTERISK-23524: testsuite: nominal DISA tests: n/p options |
ASTERISK-23525: testsuite: off-nominal DISA tests: bad authentication |
ASTERISK-23526: testsuite: off-nominal DISA tests: bad extension |
ASTERISK-23534: testsuite: Write a test for PJSIPQualify AMI action |
ASTERISK-23537: PJSIP DTMF INFO does not send any event or response for certain characters |
ASTERISK-23538: testsuite: directory fixes to prevent untracked files from being created in the svn directory |
ASTERISK-23539: Crash when attempting to dial from a PJSIP endpoint |
ASTERISK-23540: Attended transfer stops call recordings on outbound calls |
ASTERISK-23541: Asterisk 12.1.0 Not respecting directmedia=no and issuing REINVITE |
ASTERISK-23542: Dynamic realtime SIP peers using callbackextension will not register after a 'sip reload' where sip.conf has changed |
ASTERISK-23543: testsuite: Fix the sip one legged transfer test failures |
ASTERISK-23545: Confbridge talker detection settings configuration load bug |
ASTERISK-23546: CB_ADD_LEN does not do what you'd think |
ASTERISK-23547: [patch] app_queue removing callers from queue when reloading |
ASTERISK-23548: POST to ARI sometimes returns no body on success |
ASTERISK-23549: Intermittent Blind Transfer faliure when using AGI |
ASTERISK-23550: Newer sound sets don't show up in menuselect |
ASTERISK-23552: http: support persistent connections |
ASTERISK-23553: Add ast_spinlock capability to lock.h |
ASTERISK-23554: [patch]deadlock on forced disconnect of DAHDI PRI span |
ASTERISK-23556: Compilation warning for invert.c (array subscript is above array bounds) |
ASTERISK-23557: HEP/PJSIP: Add modules to support integrating Homer with PJSIP |
ASTERISK-23558: UPDATE causes chan_pjsip channel to be taken off hold |
ASTERISK-23559: app_voicemail fails to load after fix to dialplan functions |
ASTERISK-23560: [ARI] MOH doesn't indicate progress |
ASTERISK-23562: testsuite: Write a fast picture update test |
ASTERISK-23564: [patch]TLS/SRTP status of channel not currently available in a CLI command |
ASTERISK-23568: SendDTMF fails for repeating number |
ASTERISK-23569: Spanish translation for digits/h-billion is wrong |
ASTERISK-23570: CLI config reload returns 'No such module 'core'' for modules that appear to be independent - documentation needed or bug? |
ASTERISK-23571: 'logger set level' CLI commands don't toggle logger message types on or off |
ASTERISK-23572: chan_pjsip call from endpoint using WS transport to endpoint using UDP sometimes results in a crash after a call from res_rtp_asterisk into pjproject |
ASTERISK-23573: Crash when transferring unbridged call - in bridge_app_subscribed at stasis/app.c |
ASTERISK-23576: Build failure on SmartOS / Illumos / SunOS |
ASTERISK-23577: res_rtp_asterisk: Crash in ast_rtp_on_turn_rtp_state when RTP instance is NULL |
ASTERISK-23582: [patch]Inconsistent column length in *odbc |
ASTERISK-23583: pbx_realtime: Set Asterisk App - Context gets changed to value of "Set" variable |
ASTERISK-23584: PJSIP 'Unable to create channel' when attempting to call from endpoint with UDP transport to one using WebSockets |
ASTERISK-23588: ARI: Crash when unsubscribing from bridge |
ASTERISK-23590: confbridge menu doesn't accept # as definable menu option |
ASTERISK-23591: [patch]testsuite: Many chan_sip session_timer tests leave channels active |
ASTERISK-23592: [patch]Improvements to chan_unistim |
ASTERISK-23593: If the variable contains ").:." then it can not be used as a function argument |
ASTERISK-23594: Problem inserting CEL records when certain characters are used |
ASTERISK-23595: Exceptionally long voice queue length queuing |
ASTERISK-23596: Asterisk 11 Named ACLs - Could not reload ACL config |
ASTERISK-23600: rtp packet len wrong size |
ASTERISK-23605: res_http_websocket: Race condition in shutting down websocket causes crash |
ASTERISK-23607: No parameter to change confbridge encoding for recording s |
ASTERISK-23608: ControlPlayback fails to play files with names containing certain non-alpha characters |
ASTERISK-23609: Security: AMI action MixMonitor allows arbitrary programs to be run |
ASTERISK-23611: SIP registration remaining expiry time is not updated on shutdown |
ASTERISK-23612: sip registration expiry is not updated on shutdown |
ASTERISK-23613: Endpoint identification fails when user part of From header SIP-URI contains a tel URI style parameter |
ASTERISK-23614: chan_pjsip doesn't have equivalent to chan_sip 'usereqphone' for setting ';user=phone' parameter in URIs |
ASTERISK-23615: logger reload causes asterisk to crash |
ASTERISK-23616: Big memory leak in logger.c |
ASTERISK-23617: Asterisk segfault when AgentLogin concur in a time frame using Realtime ODBC |
ASTERISK-23619: Call negotiating iLBC results in speex error messages and one way audio |
ASTERISK-23620: Code path in app_stack fails to unlock list |
ASTERISK-23623: chan_sip deadlocks with TCP connections |
ASTERISK-23624: Add silence suppression to chan_sip |
ASTERISK-23625: Function needed to enumerate sip headers |
ASTERISK-23626: CDR custom field is not inserted to DB when call is not answered |
ASTERISK-23627: Asterisk CPU usage is 100% |
ASTERISK-23628: chan_unistim.c: 2 * bad if tests |
ASTERISK-23629: PJ ICE Rx error status code: 370400 'Bad Request'. |
ASTERISK-23630: chan_sip warnings 'Failure to write to tcp/tls socket' along with failed registrations |
ASTERISK-23634: With TURN Asterisk crashes on multiple (7-10) concurrent WebRTC (avpg/encryption/icesupport) calls |
ASTERISK-23635: Crash in framehook during ast_write when called by app_swift |
ASTERISK-23636: [patch]Filesystem based dynamic MoH classes |
ASTERISK-23637: Missinterpretation of DTMF from GXP-2200 in res_rtp_asterisk.c |
ASTERISK-23639: PJSIP Realtime: Alembic migration needed in order to widen some string columns |
ASTERISK-23641: testsuite: Fix rest_api/bridges/attended_transfer test failures |
ASTERISK-23643: PJSIP Testing: Nominal local attended transfers (Hold with REFER/Replaces) |
ASTERISK-23644: PJSIP Testing: Nominal local attended transfers (Direct Media, Hold, REFER/Replaces) |
ASTERISK-23649: [patch]Support for DTLS retransmission |
ASTERISK-23650: Intermittent segfault in string functions |
ASTERISK-23651: Reloading some modules that are loaded already, results in 'No such module' before a successful reload |
ASTERISK-23653: Crash when acknowledging timer in bridge_softmix in Asterisk 1.8 |
ASTERISK-23654: Add 'pjsip reload' to default cli_aliases.conf |
ASTERISK-23656: Playbak afte recording fails,but file is there |
ASTERISK-23657: Monitor: Kill with Fire. |
ASTERISK-23660: Add priority escalation to queues |
ASTERISK-23661: Queue global priorities |
ASTERISK-23662: Extended Wrap terminated by AMI |
ASTERISK-23663: Dynamically set MOH for queue |
ASTERISK-23664: Incorrect H264 specification in SDP. |
ASTERISK-23665: Wrong mime type for codec H263-1998 (h263+) |
ASTERISK-23666: CLONE - nested functions aren't portable |
ASTERISK-23667: features.conf.sample is unclear as to which options can or cannot be set in the general section |
ASTERISK-23671: PRESENCE_STATE write base64 option not working as intended |
ASTERISK-23672: PJSIP Digium presence notifications are not sent if only the subtype or message changes |
ASTERISK-23673: Security: DOS by consuming the number of allowed HTTP connections. |
ASTERISK-23675: [patch] Segmentation Fault on first SIP registration using res_config_odbc |
ASTERISK-23676: SIP Channel driver emits autodestruct WARNING with owner in place: channel reference leak |
ASTERISK-23677: [res_odbc] Syntax error in peer registration |
ASTERISK-23678: ARI: write a test for /asterisk/info |
ASTERISK-23679: [patch]CDR userfield merged incorrectly in ast_bridge_call |
ASTERISK-23680: Crash in hangup, cdr, sqlite |
ASTERISK-23681: Monitor() execution failed after call completed |
ASTERISK-23682: kernel: asterisk[7122]: segfault at 0000000000000018 rip 000000365fc7b334 rsp 00002acf76c7f3d8 error 4 |
ASTERISK-23683: #includes - wildcard character in a path more than one directory deep - results in no config parsing on module reload |
ASTERISK-23688: Asterisk crashes when writing frame in framehook in app_swift |
ASTERISK-23689: CLONE - SIP reload does nothing - Explanation why |
ASTERISK-23690: Can't delete voicemail randomly |
ASTERISK-23692: ARI: Add a Messaging Capability |
ASTERISK-23695: PJSIP Testing: Nominal local attended transfers (Application) |
ASTERISK-23696: PJSIP Testing: Off-nominal local attended transfer (bad replaces) |
ASTERISK-23699: PJSIP Testing: Nominal remote attended transfers |
ASTERISK-23701: PJSIP Testing: Off-nominal remote destination attended transfers |
ASTERISK-23702: PJSIP Testing: Nominal remote target attended transfers |
ASTERISK-23705: PJSIP: Add outbound subscriptions and inbound NOTIFY request handling (MWI, Device State) |
ASTERISK-23707: Realtime Contacts: Apparent mismatch between PGSQL database state and Asterisk state |
ASTERISK-23709: Regression in Dahdi/Analog/waitfordialtone |
ASTERISK-23711: Asterisk does not compile when HAVE_PJ_TRANSACTION_GRP_LOCK is not defined |
ASTERISK-23713: Voicemail on FS overwrites last message when last index = MSGLIMIT |
ASTERISK-23715: Media Formats: Review API and tweak naming |
ASTERISK-23717: Segfault when loading XML documentation |
ASTERISK-23718: res_pjsip_incoming_blind_request: crash with NULL session channel |
ASTERISK-23719: Asterisk locks, UDP buffer overflow, 1000+ spawns of 'chan_iax2.c find_idle_thread()' |
ASTERISK-23721: Calls to PJSIP endpoints with video enabled result in leaked RTP ports |
ASTERISK-23723: Asterisk and SIP Presence |
ASTERISK-23724: Agent stay "In use" after logoff |
ASTERISK-23726: AEL crash with while, switch and continue, (in add_extensions at ael/pval.c ) |
ASTERISK-23727: Async AGI loop breaks when trying to stream missing file |
ASTERISK-23730: [patch] Blind transfers are not logged into queue_log |
ASTERISK-23731: wrong entity attribute in all NOTIFYs for device/extension state notifications - should be publisher instead of subscriber |
ASTERISK-23733: 'reload acl' fails if acl.conf is not present on startup |
ASTERISK-23734: ARI crash when using bridges with no names |
ASTERISK-23735: Transcoding makes bad choice in high-rate translations |
ASTERISK-23736: Async AGI loop breaks when AGI command returns FAILURE |
ASTERISK-23737: application_name support in cdr_pgsql and res_config_pgsql |
ASTERISK-23738: Allow the Async AGI loop to continue even in case of AGI command returning FAILURE |
ASTERISK-23739: [patch]Segfault forwarding voicemail with ODBC storage enabled and realtime voicemail_data is used |
ASTERISK-23741: CDR answer field takes value from previous field |
ASTERISK-23742: res_http_websocket: Create a websocket client |
ASTERISK-23745: Call file problem, DelayedRetry/retrying spite MaxRetries: 0 |
ASTERISK-23747: [patch]DTMF Emulation duration calculation wrong when using RTP |
ASTERISK-23748: channel ooh323 hang up calls after 30 seconds |
ASTERISK-23749: Wiki Documentation - Create a basic formatting guideline to promote consistency |
ASTERISK-23751: Wiki Documentation - Review and update of Security Event Framework content |
ASTERISK-23754: [patch] Use var/lib directory for log file configured in asterisk.conf |
ASTERISK-23755: SIGSEGV due to alignment bug on arm when destination callgroup/pickupgroup is set |
ASTERISK-23756: setvar directive when used in template and a child of said template, results in duplicate variable names |
ASTERISK-23757: One way audio and MOH at the Background for destination while transferring the calls |
ASTERISK-23758: 500 internal server error when answering a channel with ARI |
ASTERISK-23759: Crash when IMAP voicemail count reaches a high number of messages +250 |
ASTERISK-23764: Documentation - Enhance and fix documentation for Presence State in Asterisk |
ASTERISK-23765: RTP mishandling in chan_unistim |
ASTERISK-23766: [patch] Specify timeout for database write in SQLite |
ASTERISK-23767: [patch] Dynamic IAX2 registration stops trying if ever not able to resolve |
ASTERISK-23768: [patch] Asterisk man page contains a (new) unquoted minus sign |
ASTERISK-23769: critical crash in chan_unistim ? |
ASTERISK-23770: calldate missing from contrib/ast-db-manage/cdr |
ASTERISK-23771: chan_dahdi.so loads even if dahdi doesn't |
ASTERISK-23772: Asterisk Leaks FileDescriptor in handle_recordfile res_agi.c |
ASTERISK-23773: Asterisk 11.6Cert AGI is delaying to execute calls |
ASTERISK-23774: Wiki Documentation - States and Presence |
ASTERISK-23779: core set verbose = x not honoured in cli,conf |
ASTERISK-23780: Failing to join a bridge could leave the bridge orphaned. |
ASTERISK-23781: outgoing missing as enum from contrib/ast-db-manage/config |
ASTERISK-23785: Peer registration not updated in db if asterisk made the registration (proxy) |
ASTERISK-23786: TALK_DETECT: A dialplan function that emits talking start/stop events for AMI/ARI |
ASTERISK-23787: [patch]Update DB when SIP Registration (proxy) come from sippeers table in RealTime environment |
ASTERISK-23788: App FollowMe on Realtime |
ASTERISK-23789: [patch] - Registry counter badly dec. with Realtime |
ASTERISK-23790: [patch] - SIP From headers longer than 256 characters result in dropped call and 'No closing bracket' warnings. |
ASTERISK-23791: PJSIP use UUID in Contact header when Dial |
ASTERISK-23792: Mutex left locked in chan_unistim.c |
ASTERISK-23797: Wiki Documentation - Configuration/Core Configuration/DNS Manager |
ASTERISK-23800: Wiki Documentation - Document commands useful for Asterisk developers |
ASTERISK-23802: Security: Deadlock in res_pjsip_pubsub on transaction timeout |
ASTERISK-23803: AMI action UpdateConfig EmptyCat clears all categories but the requested one |
ASTERISK-23804: Asterisk Manager API Action Originate / OriginateResponse |
ASTERISK-23805: Dialplan pattern matching error |
ASTERISK-23806: Classical ACL not working with sippeers in realtime |
ASTERISK-23807: PJSip dynamic realtime registrations - ps_registrations shouldn't exist |
ASTERISK-23809: IAX2 registration fails after temporary dns failure |
ASTERISK-23810: Crash - segfault in ast_channel_hangupcause_set |
ASTERISK-23811: Improve performance of Asterisk by reducing the number of channel snapshots created |
ASTERISK-23812: One Way Audio on REFER with Jitterbuffer On |
ASTERISK-23813: DNS Manager (dnsmgr) does not restore port |
ASTERISK-23814: No call started after peer dialed |
ASTERISK-23815: confbridge sound conf-leaderhasleft not played to end_marked or wait_marked users |
ASTERISK-23816: Confbridge Message when Muting a WaitMarked User is Confusing |
ASTERISK-23818: PBX_Lua: after asterisk startup module is loaded, but dialplan not available |
ASTERISK-23819: #include in queue.conf doesn't work |
ASTERISK-23821: AMI action 'Queues' does not return a formatted 'Response:' or 'Event:' messages. Doesn't return ActionID. |
ASTERISK-23822: AMI Originate |
ASTERISK-23823: [patch] Option to keep queuerules in realtime |
ASTERISK-23824: ConfBridge: Users cannot be muted via CLI or AMI when waiting to enter a conference |
ASTERISK-23825: Alembic scripts - table queue_members missing unique index on column uniqueid |
ASTERISK-23827: autoservice thread doesn't exit at shutdown |
ASTERISK-23828: pjsip - Need a command to list active SIP subscriptions |
ASTERISK-23830: menuselect fails to read config when all resource modules are disabled |
ASTERISK-23831: Wiki Documentation - Document PJSIP configuration for SIP presence subscriptions |
ASTERISK-23832: Asterisk realtime peers |
ASTERISK-23834: res_rtp_asterisk debug message gives wrong length if ICE |
ASTERISK-23835: crash: Segfault in manager during lookup of action "Login" in action_find |
ASTERISK-23836: JACK_HOOK with more than 8Khz |
ASTERISK-23837: Can't read responses from AGI - 'Broken Pipe' errors from utils.c - astcarefulwrite when writing to STDIN |
ASTERISK-23839: AGI - RECORD FILE - documentation doesn't describe BEEP argument |
ASTERISK-23841: DTMF atxfer doesn't set CallerID for the recall calls to the transferrer. |
ASTERISK-23844: Load of pbx_lua fails on sample extensions.lua with Lua 5.2 or greater due to addition of goto statement |
ASTERISK-23846: Unistim multilines. Loss of voice after second call drops (on a second line). |
ASTERISK-23847: Alembic voicemail script - 'recording' column should be longblob on MySQL |
ASTERISK-23848: list of variables in CHANNEL function help documentation may be missing a few |
ASTERISK-23850: Park Application does not respect Return Context Priority |
ASTERISK-23852: ARI mixing bridges should propagate linkedids. |
ASTERISK-23854: Tests/pbx/dialplan: Enhance test with While/ExitWhile tests |
ASTERISK-23855: Tests/pbx/dialplan: Enhance test with URIENCODE/URIDECODE functions |
ASTERISK-23858: 'core show hints' - presence state display is wrong, plus column missing from 'core show hint' |
ASTERISK-23864: Asterisk crashing on an outbound call over PJSIP channel tech |
ASTERISK-23865: Abstract PJSIP-specific elements away from the pubsub API |
ASTERISK-23866: RLS: Implement configuration |
ASTERISK-23867: RLS: Write an application/rlmi+xml body generator |
ASTERISK-23868: RLS: Add inbound SUBSCRIBE handling |
ASTERISK-23869: RLS: Add NOTIFY handling |
ASTERISK-23870: RLS Tests: Implement RLS Nominal tests |
ASTERISK-23871: RLS Tests: Implement RLS off-nominal tests |
ASTERISK-23872: RLS tests: Lists of Lists, nominal tests |
ASTERISK-23873: RLS tests: Lists of Lists, off-nominal tests |
ASTERISK-23874: RLS tests: Batched Notification |
ASTERISK-23889: Testsuite: Write a test for CELGenUserEvent |
ASTERISK-23890: Testsuite: Update the pbx dialplan test to cover Exec/TryExec |
ASTERISK-23894: #include in queue.conf doesn't work (2) |
ASTERISK-23895: res_pjsip_t38: T.38 Framehook prevents local native bridge |
ASTERISK-23897: [patch]Change in SETUP ACK handling (checking PI) in revision 413765 breaks working environments |
ASTERISK-23898: Testsuite: Simple nominal tests for Authenticate |
ASTERISK-23899: Testsuite: Nominal tests for Authenticate with a custom prompt |
ASTERISK-23900: Testsuite: Nominal tests for Authenticate with AstDB integration |
ASTERISK-23904: #define AST_MAX_ACCOUNT_CODE 20 causes truncation |
ASTERISK-23905: [patch]Enable Forward Secrecy (PFS) in TLS |
ASTERISK-23908: [patch]When using FEC error correction, asterisk tries considers negative sequence numbers as missing |
ASTERISK-23909: Alembic scripts - table sippeers could use a longer useragent column |
ASTERISK-23910: Testsuite: Transfer and bridge tests with errors / tracebacks |
ASTERISK-23911: URIENCODE/URIDECODE: WARNING about passing an empty string is a bit over zealous |
ASTERISK-23914: *-in.wav and *-out.wav are out of sync when using monitor application |
ASTERISK-23915: Crash in pthread_mutex_trylock called from dahdi_read |
ASTERISK-23916: [patch]SIP/SDP fmtp line may include whitespace between attributes |
ASTERISK-23917: res_http_websocket: Delay in client processing large streams of data causes disconnect and stuck socket |
ASTERISK-23918: Create script, to be included with Asterisk downloads, to spit out module support status of loaded modules |
ASTERISK-23919: Modify module structure so a module can report its supported status into Asterisk so that a list can easily be retrieved |
ASTERISK-23920: Move eid functions to utils.c, mark netsock.h deprecated |
ASTERISK-23921: refcounter.py uses excessive ram for large refs files |
ASTERISK-23922: ao2_container nodes are inconsistent REF_DEBUG |
ASTERISK-23923: UDP flows bidirectionally but one way audio on webphone |
ASTERISK-23924: Don't know of any 0x0 (Nothing) formats |
ASTERISK-23926: Asterisk locks after 'restart now' command |
ASTERISK-23927: pbx_config do incorrect matching if - happens |
ASTERISK-23928: Device state collector stop to work |
ASTERISK-23930: Call Barging/Whispering issue on SIP |
ASTERISK-23937: Syntax Error ODBC SQL |
ASTERISK-23938: pri set debug 0 span <span> disables file debugging for all spans |
ASTERISK-23939: ARI: Allow for channel subscriptions on originate |
ASTERISK-23941: ARI: Attended transfers of channels into Stasis application lose information |
ASTERISK-23943: Investigate configuration options to make extraneous Stasis messages optional |
ASTERISK-23947: ActionID missing from AMI PJSIP events (PJSIPShowEndpoints, etc.) |
ASTERISK-23948: REF_DEBUG fails to record ao2_ref against objects that were already freed |
ASTERISK-23949: Bridge/0xb75254f-in Bridge/0xb75254f-out channels did not hanguped after call hanguped |
ASTERISK-23950: DYNAMIC_FEATURES maximum length |
ASTERISK-23951: Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded |
ASTERISK-23952: Realtime sippers postgresql port not integer |
ASTERISK-23953: Testsuite: Off-nominal Authenticate test |
ASTERISK-23955: sip_setoption mis-types calls to ast_rtp_instance_set_write_format and ast_rtp_instance_set_read_format |
ASTERISK-23957: Media format improvements: implement attribute caching |
ASTERISK-23958: Media format improvements: make chan_iax2 work |
ASTERISK-23959: Media format improvements: verify chan_sip works |
ASTERISK-23960: Media format improvements: verify chan_pjsip works |
ASTERISK-23966: Media format improvements: verify chan_dahdi works |
ASTERISK-23968: Function REALTIME_STORE cannot be read |
ASTERISK-23969: SendMessage AMI action Cant Send Text Message Over PJSIP |
ASTERISK-23971: In pjsip.conf, match property of identify section does not allow subnet masks |
ASTERISK-23972: [patch]sip.conf progressinband=never does not mean 'never'. |
ASTERISK-23973: testsuite: Add test rest_api/channels/ring |
ASTERISK-23974: testsuite: Add test rest_api/channels/hold nominal test |
ASTERISK-23975: Description of variables field for userEvent operation missing details. |
ASTERISK-23979: IAX2 trunk dial status and hangup cause not sent to SIP endpoint |
ASTERISK-23982: CHANNEL(pjsip,local_addr) empty on UDP transport |
ASTERISK-23983: IAX2 Potentially Causing Multiple Locks and Hanging Asterisk |
ASTERISK-23984: Infinite loop possible in ast_careful_fwrite() |
ASTERISK-23985: PresenceState Action response does not contain ActionID; duplicates Message Header |
ASTERISK-23986: Voicemail msg_id is changed when a voicemail is played or moved. |
ASTERISK-23987: BridgeWait: channel entering into holding bridge that is being destroyed fails to successfully join the newly created holding bridge |
ASTERISK-23989: [patch]SDP offer/answer fails if crypto keys added to non-crypto offer |
ASTERISK-23990: [patch] app_voicemail crash when using IMAP greetings and forwarding messages |
ASTERISK-23991: [patch]asterisk.pc file contains a small error in the CFlags returned |
ASTERISK-23992: IAX peering fails without sufficient logging if res_timing_pthread.so is not loaded |
ASTERISK-23993: After creating a channel using ari.post, ari.get returns nothing in the form of {} |
ASTERISK-23994: res_pjsip_sdp_rtp: owner address in SDP may not be fully qualified domainname |
ASTERISK-23996: No core dumps because of res_musiconhold chdir. |
ASTERISK-23997: chan_sip: port incorrectly incremented for RTCP ICE candidates in SDP answer |
ASTERISK-23998: channel.c: Exceptionally long voice queue length queuing to Local.... |
ASTERISK-23999: (Memory leak) Realtime peers are never unref |